Hi Jigar,
I use visual dialplan too. Nice tool.
Here you can find some dial plan examples and tutorials that may help you:
codezone.apstel.com
Nile
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nile Kaledon
Sent: Monday, October 25, 2010 12:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dial plan help
Hi Jigar,
I use visual dialplan too
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Nile Kaledon
*Sent:* Monday, October 25, 2010 12:06 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Dial plan help
Hi Jigar,
I use visual dialplan too. Nice tool.
Here you can find
Chapters 4, 5 and 6 is a good start.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote:
Ok Thanks Guys.
Can you guyz suggest me upto which chapters orwhat are the chapters I should
cover for my requirement.
Because Its
Un-top-posting...
On 2010-10-25 2:01 PM, Jigar Joshi jiga...@gmail.com wrote:
Ok Thanks Guys.Can you guyz suggest me upto which chapters orwhat
are the chapters I should cover for my requirement.
Because Its too long book :P
On Mon, 25 Oct 2010, Zeeshan Zakaria
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, October 25, 2010 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan help
Un
Of
*Nile Kaledon
*Sent:* Monday, October 25, 2010 12:06 PM
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Dial plan help
Hi Jigar,
I use visual dialplan too. Nice tool.
Here you can find some dial plan
Hi Jigar
I am facing issue while generating a dial plan for the following case:
all caller should be asked a code to enter than All the callers should be
connected one extension.
Try DISA component, and then use MeetMe component if you want callers to go
to conference or Dial component if you
I totally agree with Steve's wise advice. One should at least give himself a
week learning asterisk fundamentals and related Linux basics before jumping
into creating dialplans or setting up Telecom systems. Asterisk's official
book's first few chapters cover all the basics which every asterisk
Hi,
I am facing issue while generating a dial plan for the following case:
all caller should be asked a code to enter than All the callers should be
connected one extension.
also tell me testing scenario :
I have pbx setup and currently I have soft phones to use as extension.
Currently I have
Jigar Joshi wrote:
Currently I have created a dial plan using vdp I tried submitting it
here but I don't know how to extract text version for the same .
After Googling a bit, I found that VDP is Visual Dial Plan for
Asterisk. Neat little application, but I doubt you'll find many if any
On Mon, 18 Oct 2010, Jigar Joshi wrote:
@Gilles here are my requirement.can you please help me .
On Mon, 18 Oct 2010, Steve Edwards wrote:
Are you putting this out to bid or are you just too lazy to read ATFOT
(http://downloads.oreilly.com/books/9780596510480.pdf)?
On Sat, 23 Oct 2010,
: [asterisk-users] Dial Plan Help
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like
On Sun, Aug 24, 2008 at 8:11 AM, Jon Weisman [EMAIL PROTECTED] wrote:
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send
John,
This is the default behaviour anyway. If Dial() is successful,
execution of subsequent priorities in the dial plan for that extension
is not resumed. It'll only fall through to the other priorities if
Dial() fails.
I do, however, suggest supplying a timeout argument to your Dial()s.
PROTECTED] On Behalf Of Sydney Web
Hosting
Sent: July 6, 2008 8:33 PM
To: Asterisk Users List
Subject: [asterisk-users] dial plan help.
I have a question about the following dial plan.
Ring main number
playback message
If press 1 got to support
if press 2 go to sales
//Support
Play message your
I have a question about the following dial plan.
Ring main number
playback message
If press 1 got to support
if press 2 go to sales
//Support
Play message your call is important to us then ring the phone and I
pickup.
//Sales
Play message your call is important to us then ring the phone and I
So how do we set it up if I'm out of the office, or on the mobile phone and
can't answer the call.
How does it know to go to voice mail?
You set it to ring for a certain duration then go to voicemail after n seconds.
You'll want an incoming call to go to a context at which point you can start
I have the following context in the dial plan.
in extension.conf
[default]
1) context1
2) context2
3) context1,1,Macro(a)
4) context2,1,NoOp
5) context2,2,NoOp
[macro-a]
6) exten = s,1, NoOp
7) exten = s,2, MacroExit
As I expect the route of a call is 3,6,7,4,5. However, when I execute
the
-Commercial Discussion
Onderwerp: [Asterisk-Users] Dial Plan Help
All,
I've got a problem here. We are using a Digium 4 T-1 board in our * server.
The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
that when someone dials our DID it goes to the s extension and starts
playing
All,
I've got a problem here. We are using a Digium 4 T-1 board in our * server.
The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
that when someone dials our DID it goes to the s extension and starts
playing the ivr which is fine, but if someone dials an extension for
On Fri, 2004-12-03 at 15:52 -0500, [EMAIL PROTECTED] wrote:
All,
I've got a problem here. We are using a Digium 4 T-1 board in our * server.
The T-1's are ISDN. The problem I'm having is that we have an ivr setup so
that when someone dials our DID it goes to the s extension and starts
exten=200,Goto(office,102,1);forward to 102 in office context
exten=201,Goto(office,110,1);forward to 110 in office context
These are invalid -- no priority -- and hence dropped. Didn't you see the
errors while loading (it's easy to miss, there's plenty of stuff output).
Change to:
printed, so it won't be possible to change now.
Thanks,
Jon
- Original Message -
From: Luki [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, December 03, 2004 4:01 PM
Subject: Re: [Asterisk-Users] Dial Plan Help
exten=200
extension.
Watch that digittimeout..it helps.
-Matthew
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, December 03, 2004 2:52 PM
Subject: Re: [Asterisk-Users] Dial Plan Help
On Fri
Jon,
* scans through the valid extension in the context every time the user enter
a digit. If you only have single digit extensions, dialing 2 is definitive
and * can jump to that extension without waiting for further digits. But if
you have 2 and 200 defined, a single 2 is ambiguous and asterisk
On Fri, 2004-12-03 at 16:12 -0500, [EMAIL PROTECTED] wrote:
Thanks Luki,
I can't believe I overlooked that. It's working now.
Steven,
It wasn't working because I overlooked the priority. However is it still a
bad idea to have ivr and extensions beginning with the same number or within
I wish to have outgoing calls try to use a SIP/IAX provider and if this
fails, then fall back to PSTN and I am not sure how the dial plan should
look.
Can someone please post a sample of how it should look.
Thanks in advance,
Simon Brown
___
Brown
Sent: 07 June 2004 07:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dial plan help
I wish to have outgoing calls try to use a SIP/IAX provider and if this
fails, then fall back to PSTN and I am not sure how the dial plan should
look.
Can someone please post a sample of how it should look
is reached after dial times out.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon Brown
Sent: 07 June 2004 07:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dial plan help
I wish to have outgoing calls try to use a SIP/IAX provider and if this
fails
I would use:
exten = _NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten = _NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _NXXNXX,3,Congestion
exten = _NXXNXX,102,1,Busy
exten = _NXXNXX,103,1,Busy
That way if number you dial is busy it will not immediately try dialing
the same
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